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Code Issues 1 Projects Releases Wiki Activity

android, mp-webrtc-sendrecv, sendonly: cleanup

Browse source 
webrtc-unidirectional-h264.c: removed empty lines

android: removed unused var
This commit is contained in:
Costa Shulyupin 2020-04-14 20:13:37 +03:00 committed by Mathieu Duponchelle
parent 133a1593ee
commit 8c4345da7d
3 changed files with 1 additions and 15 deletions
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1
webrtc/android/app/src/main/jni/webrtc.c
View file

@ -280,7 +280,6 @@ on_offer_created (GstPromise * promise, WebRTC * webrtc)
{ {
GstWebRTCSessionDescription *offer = NULL; GstWebRTCSessionDescription *offer = NULL;
const GstStructure *reply; const GstStructure *reply;
gchar *desc;
g_assert (webrtc->app_state == PEER_CALL_NEGOTIATING); g_assert (webrtc->app_state == PEER_CALL_NEGOTIATING);

3
webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
View file

@ -404,7 +404,6 @@ incoming_call_from_peer (const gchar * peer_id)
#define STR(x) #x #define STR(x) #x
#define RTP_CAPS_OPUS(x) "application/x-rtp,media=audio,encoding-name=OPUS,payload=" STR(x) #define RTP_CAPS_OPUS(x) "application/x-rtp,media=audio,encoding-name=OPUS,payload=" STR(x)
#define RTP_CAPS_VP8(x) "application/x-rtp,media=video,encoding-name=VP8,payload=" STR(x)
static gboolean static gboolean
start_pipeline (void) start_pipeline (void)
@ -902,7 +901,6 @@ check_plugins (void)
{ {
int i; int i;
gboolean ret; gboolean ret;
GstPlugin *plugin;
GstRegistry *registry; GstRegistry *registry;
const gchar *needed[] = { "opus", "nice", "webrtc", "dtls", "srtp", const gchar *needed[] = { "opus", "nice", "webrtc", "dtls", "srtp",
"rtpmanager", "audiotestsrc", NULL "rtpmanager", "audiotestsrc", NULL
@ -911,6 +909,7 @@ check_plugins (void)
registry = gst_registry_get (); registry = gst_registry_get ();
ret = TRUE; ret = TRUE;
for (i = 0; i < g_strv_length ((gchar **) needed); i++) { for (i = 0; i < g_strv_length ((gchar **) needed); i++) {
GstPlugin *plugin;
plugin = gst_registry_find_plugin (registry, needed[i]); plugin = gst_registry_find_plugin (registry, needed[i]);
if (!plugin) { if (!plugin) {
g_print ("Required gstreamer plugin '%s' not found\n", needed[i]); g_print ("Required gstreamer plugin '%s' not found\n", needed[i]);

12
webrtc/sendonly/webrtc-unidirectional-h264.c
View file

@ -11,14 +11,10 @@
#include <json-glib/json-glib.h> #include <json-glib/json-glib.h>
#include <string.h> #include <string.h>
#define RTP_PAYLOAD_TYPE "96" #define RTP_PAYLOAD_TYPE "96"
#define SOUP_HTTP_PORT 57778 #define SOUP_HTTP_PORT 57778
#define STUN_SERVER "stun.l.google.com:19302" #define STUN_SERVER "stun.l.google.com:19302"
typedef struct _ReceiverEntry ReceiverEntry; typedef struct _ReceiverEntry ReceiverEntry;
ReceiverEntry *create_receiver_entry (SoupWebsocketConnection * connection); ReceiverEntry *create_receiver_entry (SoupWebsocketConnection * connection);
@ -48,9 +44,6 @@ static gchar *get_string_from_json_object (JsonObject * object);
gboolean exit_sighandler (gpointer user_data); gboolean exit_sighandler (gpointer user_data);
struct _ReceiverEntry struct _ReceiverEntry
{ {
SoupWebsocketConnection *connection; SoupWebsocketConnection *connection;
@ -59,8 +52,6 @@ struct _ReceiverEntry
GstElement *webrtcbin; GstElement *webrtcbin;
}; };
const gchar *html_source = " \n \ const gchar *html_source = " \n \
<html> \n \ <html> \n \
<head> \n \ <head> \n \
@ -166,9 +157,6 @@ const gchar *html_source = " \n \
</html> \n \ </html> \n \
"; ";
ReceiverEntry * ReceiverEntry *
create_receiver_entry (SoupWebsocketConnection * connection) create_receiver_entry (SoupWebsocketConnection * connection)
{ {